Théo Monnom
33dfcb27b0
- Fix datachannel deadlock on dispose - Fix audio device being disposed on the wrong thread - Allow multiple RtcRuntime to be created (Useful for unit tests where we use multiple PeerConnectionFactory)
117 lines
3.7 KiB
C++
117 lines
3.7 KiB
C++
/*
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* Copyright 2023 LiveKit
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*
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* Licensed under the Apache License, Version 2.0 (the “License”);
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an “AS IS” BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#pragma once
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#include <memory>
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#include "api/media_stream_interface.h"
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#include "api/rtp_receiver_interface.h"
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#include "api/rtp_sender_interface.h"
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#include "livekit/helper.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/physical_socket_server.h"
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#include "rtc_base/ssl_adapter.h"
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#include "rust/cxx.h"
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#ifdef WEBRTC_WIN
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#include "rtc_base/win32_socket_init.h"
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#endif
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namespace livekit {
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class RtcRuntime;
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class LogSink;
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} // namespace livekit
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#include "webrtc-sys/src/webrtc.rs.h"
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namespace livekit {
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class MediaStreamTrack;
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class RtpReceiver;
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class RtpSender;
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// Using a shared_ptr in RtcRuntime allows us to keep a strong reference to it
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// on resources that depend on it. (e.g: AudioTrack, VideoTrack).
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class RtcRuntime : public std::enable_shared_from_this<RtcRuntime> {
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public:
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[[nodiscard]] static std::shared_ptr<RtcRuntime> create() {
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return std::shared_ptr<RtcRuntime>(new RtcRuntime());
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}
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RtcRuntime(const RtcRuntime&) = delete;
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RtcRuntime& operator=(const RtcRuntime&) = delete;
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~RtcRuntime();
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rtc::Thread* network_thread() const;
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rtc::Thread* worker_thread() const;
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rtc::Thread* signaling_thread() const;
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std::shared_ptr<MediaStreamTrack> get_or_create_media_stream_track(
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rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track);
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std::shared_ptr<AudioTrack> get_or_create_audio_track(
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rtc::scoped_refptr<webrtc::AudioTrackInterface> track);
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std::shared_ptr<VideoTrack> get_or_create_video_track(
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rtc::scoped_refptr<webrtc::VideoTrackInterface> track);
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private:
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RtcRuntime();
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std::unique_ptr<rtc::Thread> network_thread_;
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std::unique_ptr<rtc::Thread> worker_thread_;
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std::unique_ptr<rtc::Thread> signaling_thread_;
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// Lists used to make sure we don't create multiple wrappers for one
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// underlying webrtc object. (e.g: webrtc::VideoTrackInterface should only
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// have one livekit::VideoTrack associated with it).
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// The only reason we to do that is to allow to add states inside our
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// wrappers (e.g: the sinks_ member inside AudioTrack)
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// DataChannel and the PeerConnectionFactory don't need to do this (There's no
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// way to retrieve them after creation)
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webrtc::Mutex mutex_;
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std::vector<std::weak_ptr<MediaStreamTrack>> media_stream_tracks_;
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// We don't have additonal state in RtpReceiver and RtpSender atm..
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// std::vector<std::weak_ptr<RtpReceiver>> rtp_receivers_;
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// std::vector<std::weak_ptr<RtpSender>> rtp_senders_;
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#ifdef WEBRTC_WIN
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// rtc::WinsockInitializer winsock_;
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// rtc::PhysicalSocketServer ss_;
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// rtc::AutoSocketServerThread main_thread_{&ss_};
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#endif
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};
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class LogSink : public rtc::LogSink {
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public:
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LogSink(rust::Fn<void(rust::String message, LoggingSeverity severity)> fnc);
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~LogSink();
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void OnLogMessage(const std::string& message,
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rtc::LoggingSeverity severity) override;
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void OnLogMessage(const std::string& message) override {}
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private:
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rust::Fn<void(rust::String message, LoggingSeverity severity)> fnc_;
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};
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std::unique_ptr<LogSink> new_log_sink(
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rust::Fn<void(rust::String, LoggingSeverity)> fnc);
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rust::String create_random_uuid();
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} // namespace livekit
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